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 Post subject: SIP and DID acting as same, big problem
PostPosted: Sat Feb 20, 2010 8:01 pm 
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Joined: Mon Jan 25, 2010 12:43 am
Posts: 15
Hello,

with my knowledge of VOIP i know that when people use a softphone to call they should not experience any kind of prompts. I have a DID setup as well.
The issue is my users hear prompts when they call using SIP. I can turn this off but this then also affects the DID number. Somehow they are interconnected which is completely wrong. Also when i set use_dnid to YES that is what i want for SIP but that affects DID as well. Something is wrong and i need these to work. Can someone help please? here is my extensions..,


Code:
[a2billing]
; CallingCard application
exten => _X.,1,Answer
exten => _X.,2,Wait,2
exten => _X.,3,DeadAGI(a2billing.php|-11)
exten => _X.,4,Wait,2
exten => _X.,5,Hangup

Also here is the SIP.conf that i am using

Code:
[general]
context=default      ; Default context for incoming calls
;allowguest=no         ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
            ; if asterisk was compiled with OSP support.
;realm=mydomain.tld      ; Realm for digest authentication
            ; defaults to "asterisk"
            ; Realms MUST be globally unique according to RFC 3261
            ; Set this to your host name or domain name
bindport=5060         ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0      ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes         ; Enable DNS SRV lookups on outbound calls
            ; Note: Asterisk only uses the first host
            ; in SRV records
            ; Disabling DNS SRV lookups disables the
            ; ability to place SIP calls based on domain
            ; names to some other SIP users on the Internet
            
;domain=mydomain.tld      ; Set default domain for this host
            ; If configured, Asterisk will only allow
            ; INVITE and REFER to non-local domains
            ; Use "sip show domains" to list local domains
;domain=mydomain.tld,mydomain-incoming
            ; Add domain and configure incoming context
            ; for external calls to this domain
;domain=1.2.3.4         ; Add IP address as local domain
            ; You can have several "domain" settings
;allowexternalinvites=no   ; Disable INVITE and REFER to non-local domains
            ; Default is yes
;autodomain=yes         ; Turn this on to have Asterisk add local host
            ; name and local IP to domain list.
;pedantic=yes         ; Enable slow, pedantic checking for Pingtel
            ; and multiline formatted headers for strict
            ; SIP compatibility (defaults to "no")
;tos=184         ; Set IP QoS to either a keyword or numeric val
;tos=lowdelay         ; lowdelay,throughput,reliability,mincost,none
;maxexpiry=3600         ; Max length of incoming registration we allow
;defaultexpiry=120      ; Default length of incoming/outgoing registration
;notifymimetype=text/plain   ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10         ; Default time between mailbox checks for peers
;vmexten=voicemail      ; dialplan extension to reach mailbox sets the
                  ; Message-Account in the MWI notify message
                  ; defaults to "asterisk"
;videosupport=yes      ; Turn on support for SIP video
;recordhistory=yes      ; Record SIP history by default
            ; (see sip history / sip no history)

;disallow=all
allow=g729
allow=ulaw         ; Allow codecs in order of preference
;allow=alaw
;allow=ilbc         ;
;musicclass=default      ; Sets the default music on hold class for all SIP calls
            ; This may also be set for individual users/peers
;language=en         ; Default language setting for all users/peers
            ; This may also be set for individual users/peers
;relaxdtmf=yes         ; Relax dtmf handling
;rtptimeout=60         ; Terminate call if 60 seconds of no RTP activity
            ; when we're not on hold
;rtpholdtimeout=300      ; Terminate call if 300 seconds of no RTP activity
            ; when we're on hold (must be > rtptimeout)
;trustrpid = no         ; If Remote-Party-ID should be trusted
;sendrpid = yes         ; If Remote-Party-ID should be sent
;progressinband=never      ; If we should generate in-band ringing always
            ; use 'never' to never use in-band signalling, even in cases
            ; where some buggy devices might not render it
            ; Valid values: yes, no, never Default: never
;useragent=Asterisk PBX      ; Allows you to change the user agent string
;promiscredir = no         ; If yes, allows 302 or REDIR to non-local SIP address
                             ; Note that promiscredir when redirects are made to the
                             ; local system will cause loops since SIP is incapable
                             ; of performing a "hairpin" call.
;usereqphone = no      ; If yes, ";user=phone" is added to uri that contains
            ; a valid phone number
dtmfmode = rfc2833      ; Set default dtmfmode for sending DTMF. Default: rfc2833
            ; Other options:
            ; info : SIP INFO messages
            ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
            ; auto : Use rfc2833 if offered, inband otherwise

;compactheaders = yes      ; send compact sip headers.
;sipdebug = yes         ; Turn on SIP debugging by default, from
            ; the moment the channel loads this configuration
;subscribecontext = default   ; Set a specific context for SUBSCRIBE requests
            ; Useful to limit subscriptions to local extensions
            ; Settable per peer/user also
;notifyringing = yes      ; Notify subscriptions on RINGING state
;alwaysauthreject = yes      ; When an incoming INVITE or REGISTER is to be rejected,
                ; for any reason, always reject with '401 Unauthorized'
            ; instead of letting the requester know whether there was
            ; a matching user or peer for their request
;
; If regcontext is specified, Asterisk will dynamically create and destroy a
; NoOp priority 1 extension for a given peer who registers or unregisters with
; us.  The actual extension is the 'regexten' parameter of the registering
; peer or its name if 'regexten' is not provided.  More than one regexten may
; be supplied if they are separated by '&'.  Patterns may be used in regexten.
;
;regcontext=sipregistrations
;
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
;       register => user[:secret[:authuser]]@host[:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension needs to
; be defined in extensions.conf to be able to accept calls from this SIP proxy
; (provider).
;
; host is either a host name defined in DNS or the name of a section defined
; below.
;
; Examples:
;
;register => 1234:[email protected]   
;
;     This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_proxy/1234
;
;    Register 2345 at sip provider 'sip_proxy'.  Calls from this provider
;    connect to local extension 1234 in extensions.conf, default context,
;    unless you configure a [sip_proxy] section below, and configure a
;    context.
;    Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
;    Tip 2: Use separate type=peer and type=user sections for SIP providers
;           (instead of type=friend) if you have calls in both directions
 
;registertimeout=20      ; retry registration calls every 20 seconds (default)
;registerattempts=10      ; Number of registration attempts before we give up
            ; 0 = continue forever, hammering the other server until it
            ; accepts the registration
            ; Default is 0 tries, continue forever
;callevents=no         ; generate manager events when sip ua performs events (e.g. hold)

;----------------------------------------- NAT SUPPORT ------------------------
; The externip, externhost and localnet settings are used if you use Asterisk
; behind a NAT device to communicate with services on the outside.

;externip = 200.201.202.203   ; Address that we're going to put in outbound SIP messages
            ; if we're behind a NAT

            ; The externip and localnet is used
            ; when registering and communicating with other proxies
            ; that we're registered with
;externhost=foo.dyndns.net   ; Alternatively you can specify an
            ; external host, and Asterisk will
            ; perform DNS queries periodically.  Not
            ; recommended for production
            ; environments!  Use externip instead
;externrefresh=10      ; How often to refresh externhost if
            ; used
            ; You may add multiple local networks.  A reasonable set of defaults
            ; are:
;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
;localnet=10.0.0.0/255.0.0.0   ; Also RFC1918
;localnet=172.16.0.0/12      ; Another RFC1918 with CIDR notation
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network

; The nat= setting is used when Asterisk is on a public IP, communicating with
; devices hidden behind a NAT device (broadband router).  If you have one-way
; audio problems, you usually have problems with your NAT configuration or your
; firewall's support of SIP+RTP ports.  You configure Asterisk choice of RTP
; ports for incoming audio in rtp.conf
;
;nat=no            ; Global NAT settings  (Affects all peers and users)
                                ; yes = Always ignore info and assume NAT
                                ; no = Use NAT mode only according to RFC3581
                                ; never = Never attempt NAT mode or RFC3581 support
            ; route = Assume NAT, don't send rport
            ; (work around more UNIDEN bugs)

;rtcachefriends=yes      ; Cache realtime friends by adding them to the internal list
            ; just like friends added from the config file only on a
            ; as-needed basis? (yes|no)

;rtupdate=yes         ; Send registry updates to database using realtime? (yes|no)
            ; If set to yes, when a SIP UA registers successfully, the ip address,
            ; the origination port, the registration period, and the username of
            ; the UA will be set to database via realtime. If not present, defaults to 'yes'.

;rtautoclear=yes      ; Auto-Expire friends created on the fly on the same schedule
            ; as if it had just registered? (yes|no|<seconds>)
            ; If set to yes, when the registration expires, the friend will vanish from
            ; the configuration until requested again. If set to an integer,
            ; friends expire within this number of seconds instead of the
            ; registration interval.

;ignoreregexpire=yes      ; Enabling this setting has two functions:
            ;
            ; For non-realtime peers, when their registration expires, the information
            ; will _not_ be removed from memory or the Asterisk database; if you attempt
            ; to place a call to the peer, the existing information will be used in spite
            ; of it having expired
            ;
            ; For realtime peers, when the peer is retrieved from realtime storage,
            ; the registration information will be used regardless of whether
            ; it has expired or not; if it expires while the realtime peer is still in
            ; memory (due to caching or other reasons), the information will not be
            ; removed from realtime storage

; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
; domains, each of which can direct the call to a specific context if desired.
; By default, all domains are accepted and sent to the default context or the
; context associated with the user/peer placing the call.
; Domains can be specified using:
; domain=<domain>[,<context>]
; Examples:
; domain=myasterisk.dom
; domain=customer.com,customer-context
;
; In addition, all the 'default' domains associated with a server should be
; added if incoming request filtering is desired.
; autodomain=yes
;
; To disallow requests for domains not serviced by this server:
; allowexternaldomains=no

; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
                          ; non-peers, use your primary domain "identity"
                          ; for From: headers instead of just your IP
                          ; address. This is to be polite and
                          ; it may be a mandatory requirement for some
                          ; destinations which do not have a prior
                          ; account relationship with your server.


sorry for all the extra writing. Please can someone help me on how i can seperate these so they work independently of each other?

Thanks


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 Post subject: Re: SIP and DID acting as same, big problem
PostPosted: Mon Feb 22, 2010 9:04 am 
Offline

Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Add a second agi-conf, and use the dialplan to select the appropriate agi-conf.

e.g.
Code:
[a2billing]
;Uses agiconf1 for VoIP accounts.
exten => _X.,1,DeadAGI(a2billing.php|1)
exten => _X.,n,Hangup

[a2billing-callthrough]
;Uses agiconf2 for call through accounts.
exten => _X.,1,DeadAGI(a2billing.php|2)
exten => _X.,n,Hangup


Joe


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